Without limiting the scope of the invention, its background is described in connection with a scheme for decoding a composite frequency modulation (FM) stereo signals, as an example.
A composite FM Stereo signal is of the form: EQU fm(t)=[l(t)+r(t)]+A.sub.p sin (.omega..sub.p t)+[l(t)-r(t)] sin (2.omega..sub.p t) (1)
where:
fm(t) is the time varying value of the composite signal; PA1 l(t) is the time varying value of the left channel signal; PA1 r(t) is the time varying value of the right channel signal; PA1 A.sub.p is the amplitude of the 19 Khz pilot signal; PA1 .omega..sub.p is the pilot frequency of 2II *19K radians per second (19 Khz).
FIG. 1 illustrates a frequency spectrum of a typical FM stereo composite signal showing the components of Equation 1. The components include a sum of the left and right channel signals covering a 15 Khz bandwidth from DC to 15 KHz and the difference of the left and right channels modulated to and centered about a 38 Khz suppressed stereo subcarrier signal, with upper and lower sidebands spanning a 30 KHz bandwidth. Additionally the composite signal fm(t) signal includes a 19 KHz tone signal, commonly referred to as the pilot signal which is used as a reference signal for the radio receiver. The composite signal may also contain subsidiary signals in the 53 Khz to 75 KHz bandwidth, such as a subsidiary communication authorization (SCA) signal. These signals are excluded from FIG. 1 for clarity.
The composite signal fm(t) must be separated into left and right channels ("decoded") in order to reproduce the broadcast message in stereo. This requires extracting from the composite signal the values of the left channel and the right channel signals, l(t) and r(t) respectively of Equation 1 in isolation from the other components of the composite signal.
One analog method of decoding involves first passing the composite stereo through a low pass filter to remove the subsidiary signals, such as the SCA signal, leaving only the basic stereo signal. The basic stereo signal is then mixed with the 38 KHz subcarrier with one resulting component being one sideband of the (L-R) signal translated down to baseband. The pre-mixing basic stereo signal and the stereo signal mixed with the stereo subcarrier are in parallel passed through a low pass filter and then to a summing circuit and a subtracting circuit, the summing circuit adding the two signals together and the subtracting circuit subtracting the mixed signal from the unaltered basic stereo signal. One of the resulting components from the summation is 2R(t) (i.e. twice the time varying value of the right channel component). As a result of the subtraction, one of the resulting components is 21(t) (i.e. twice the time varying value of the left channel component). The right and left channel information is then easily extracted by filtering out the remaining components resulting from the mixing, summation and subtraction operations. This analog approach is well known in the art and is susceptible to all of the disadvantages inherent with analog signal processing such, problems with noise, drift with temperature, and overall circuit complexity.
One digital decoding approach which overcomes the disadvantages inherent with analog decoding circuitry involves converting from the analog to the digital domain the composite FM stereo signal output from the FM discriminator. In this instance, the 38 Khz modulated portions of composite signal are sampled at selected points when the term [sin2.omega..sub.p ] (or alternatively sin.omega..sub.sc, where .omega..sub.sc is the angular frequency of the subcarrier, typically 38 KHz) in Equation 1 is equal to plus or minus one (the ninety degree points on the stereo subcarrier) such that the composite signal is equal to either twice the left channel (21) or twice the right channel (2r). The left and right channel information can then be easily extracted. The substantial difficulty with this approach is that, if the samples vary from the ninety degree points on the subcarrier, the sine of the subcarrier signal will not equal plus or minus one and a given sample will not represent a signal which is essentially purely right channel information or purely left channel information; the result is a deterioration in channel separation. One means of overcoming this problem is to use a voltage controlled oscillator feedback path to phase lock the sampling frequency to the pilot signal frequency. The 19 KHz pilot signal is then used to determine when sampling of the 38 KHz modulated information will occur. This method however requires substantially complex and costly hardware to implement.
Thus, the need has arisen for improved devices, systems and methods for decoding composite signals. Such devices, systems and methods would overcome the substantial technical disadvantages inherent with currently available analog decoding means and the substantial cost and complexity disadvantages inherent with currently available digital decoding means.